ZyXEL P-2302R Bedienungsanleitung PDF herunterladen (Seite 109)

P-2302R Series User’s Guide

Chapter 8 VoIP Screens 108

Server Address Your VoIP service provider must host a STUN server in order for you to use STUN.

Type the IP address or domain name (up to 127 ASCII characters) of the STUN

server in this field.

Server Port Enter the STUN server’s listening port for STUN requests in this field. Leave this

field set to the default if your VoIP service provider did not give you a server port

number for STUN.

Use NAT

Active Check this box to use a NAT router’s public IP address and SIP port number in the

Prestige’s SIP messages. This eliminates the need for STUN or a SIP ALG. You

must also configure the NAT router to forward traffic with this port number to the

Prestige.

Server Address Enter the NAT router’s public IP address or domain name (up to 127 ASCII

characters) in this field.

Server Port Enter the port number that your SIP sessions use with the public IP address of the

NAT router.

Outbound Proxy

Active Check this box if your VoIP service provider has a SIP outbound server to handle

voice calls. This allows the Prestige to work with any type of NAT router and

eliminates the need for STUN or a SIP ALG. Turn off any SIP ALG on a NAT router

in front of the Prestige to keep it from retranslating the IP address (since this is

already handled by the outbound proxy server).

Server Address Enter the IP address or domain name (up to 127 ASCII characters) of the SIP

outbound proxy server in this field.

Server Port Enter the SIP outbound proxy server’s listening port for SIP outbound proxy

requests in this field. Leave this field set to the default if your VoIP service provider

did not give you a server port number for the SIP outbound proxy server.

NAT Keep Alive

Enable NAT Keep

Alive

You must have outbound proxy enabled to use NAT keep alive.

Enable NAT keep alive to have the Prestige send SIP notify messages to the SIP

server. Use this to keep a NAT router located between the Prestige and the SIP

server from timing out and dropping your Prestige’s SIP NAT sessions.

Keep Alive Interval Set how often (in seconds) the Prestige should send SIP notify messages to the

SIP server.

Dual-Tone-Multi-

Frequency

(DTMF)

DTMF Mode The Dual-Tone Multi-Frequency (DTMF) mode sets how the Prestige handles the

tones that your telephone makes when you push its buttons. It is recommended

that you use the same mode that your VoIP service provider uses.

Select RFC 2833 to send the DTMF tones in RTP packets.

Select PCM (Pulse Code Modulation) to include the DTMF tones in the voice data

stream. This method works best when you are using a codec that does not use

compression (like G.711). Codecs that use compression (like G.729) could distort

the tones.

Select SIP INFO to send the DTMF tones in SIP messages.

MWI (Message

Waiting Indication)

Table 23 VoIP Advanced (continued)

LABEL DESCRIPTION

1 2 ... 104 105 106 107 108 109 110 111 112 113 114 ... 383 384

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